Transmission rate control is an important part of real-time video and audio communication systems, as it enables dynamic adaptation of the systems to time varying network conditions. Transmission rate control prevents the creation of a backlog of packets at an intermediate router along the network path between the sender and the receiver. In the absence of rate control, a backlog of packets (i.e. a congestion event) is created whenever there is a mismatch between the transmission rate of the sender and the available network bit rate on the path. The congestion events can contribute to excessive delays and potential losses of transmitted packets, thereby significantly degrading the audiovisual quality of a communication session.
Decisions on transmission rate control actions can be made either at the sender, based on receiver feedback, or at the receiver, based on information associated with arriving packets. In the latter case, the decisions are then signaled back to the sender. Among the most notable the latter case, the decisions are then signaled back to the sender. Among the most notable examples of prior work on sender-driven rate control is the TCP-friendly scheme. (See e.g., “Equation-Based Congestion Control for Unicast Applications,” Sally Floyd, Mark Handley, Jitendra Padhye, and Joerg Widmer, August 2000, SIGCOMM 2000). In this scheme, an equation-based technique for congestion control is used to compute the available network bit rate based on running estimates of the round-trip time, of the packet loss, and of the variance of the round-trip time, as experienced on the network path between the sender and the receiver. The sender adjusts its transmission rate dynamically based on the computed available bit rate. In the TCP-friendly scheme, feedback based only on the last received packet within every round-trip time interval is sent to the receiver.
It is advantageous to consider rate control techniques, which differ from the TCP-friendly approach in that: (1) they are performed at the receiver, and (2) they employ a different mechanism to compute the available bit rate estimate. The former distinction is advantageous in scenarios of low-latency communication such as video conferencing, where decision-making at the receiver eliminates excessive feedback from the receiver to the sender. The latter difference can provide improvement in performance over the TCP-friendly scheme, as it may be possible to treat every arriving packet individually before computing a rate control decision. Prior art techniques related to receiver-driven rate control assume a uniform transmission rate and therefore operate on the inter-arrival times of arriving packets at the receiver. This uniform transmission rate assumption is not true in the case of video communications where the sending rate can exhibit significant variations over short periods of time, thereby rendering the analysis employed by such techniques inappropriate in video communication scenarios. Some prior art techniques related to receiver-driven rate control include the concept of monitoring the increase in inter-arrival time relative to the fixed inter packet-probe interval used by the sender. Further, the prior art techniques related to receiver-driven rate control assume time-synchronization between the sender and the receiver, so that the one-way delay can be accurately computed from the time-stamp and the arrival time associated with a packet. (See e.g., M. Jain and C. Dovrolis, “Pathload: a measurement tool for end-to-end available bandwidth,” Proc. Passive Active Measurements, Fort Collins, Colo., March 2002; V. Ribeiro, R. Riedi, R. Baraniuk, J. Navratil, and L. Cottrell, “pathChirp: efficient available bandwidth estimation for network paths,” Passive and Active Measurement Workshop 2003; and Shawn W. Smith, U.S. Pat. No. 6,996,626, “Continuous bandwidth assessment and feedback for voice-over-internet-protocol (VoIP) comparing packet's voice duration and arrival rate”).
Consideration is being given to developing mechanisms for efficient transmission rate control in video communication systems. In particular, attention is being directed toward mechanisms that operate on information associated with arriving packets at the receiver and provide the sender with an estimate of the available bit rate on the network path between the sender and the receiver. The desirable mechanisms do not assume time synchronization between the sender and the receiver to compute the one-way delay from the time-stamp and the arrival time associated with a packet. Further, the desirable mechanisms may combine concepts of monitoring the increase in inter-arrival time relative to the fixed inter packet-probe interval used by the sender with new ways to detect congestion events in the network, for example, based on the variance of the delay jitter. By their design, the desirable mechanisms will overcome many of the shortcomings of existing transmission rate control solutions.